The trend in third generation (3G), fourth generation (4G), and further advanced wireless networks is towards the all-IP network, which involves extending IP functionality and services into the radio access network (RAN) and the air-interface. Extending IP over the air-interface facilitates enhancements to client applications, devices, and end-user experience. Protocols have been developed and are being improved for providing IP services to wireless users and integrating the provisioning of IP to wireless users with the provisioning of IP to all other users, e.g., wireline, wireless local area networks (LAN), and the like. One such protocol is the IP multimedia subsystem (IMS). IMS is an architectural framework for delivering IP multimedia to mobile users based in part on session initiation protocol (SIP). Voice over IP (VoIP) is one of the key IP applications since voice still accounts for a significant percentage of the average revenue per user (ARPU) earned by most wireless operators.
VoIP services are real-time service applications. Because of the real-time nature of VoIP applications, the end-user is very sensitive to the end-to-end delay for the VoIP packets. This end-to-end delay directly affects a user's quality perception. To this end, the International Telecommunications Union (ITU) has sponsored guidelines and standards for improving the overall quality of telecommunications services. ITU G.114, for example, provides guidelines on what the end-to-end delay should be for VoIP services and also formulates methods to compute the quality of VoIP services. The end-to-end delay guidelines provided by ITU G.114 are then used to formulate the end-to-end delay criteria for VoIP for wireless networks.
Transmitting VoIP services over wireless networks creates new issues that have begun to be addressed. FIG. 1 is a block diagram illustrating a typical wireless network connecting two access terminals, ATs 100 and 106. In basic operation, the VoIP transmission originates at AT 100 and is received at base transceiver (BTS) 101. BTS 101 sends the VoIP transmission to the originating network control 102, which may comprises network components such as the base station controller (BSC), packet data service node (PDSN), and the like. After processing at originating network control 102, the VoIP transmission is transmitted over transmission network 103. Terminating network control 104 receives the VoIP transmission from transmission network 103 when the target receiver is within that network. When the location and address information is determined by terminating network control 104, the VoIP transmission is sent to BTS 105 for scheduling delivery to the target user, AT 106.
One of the key segments in this wireless communication process lies at the air interfaces, air interface 107 and 108. The air interface is the communication interface between the access terminal and the BTS. Wireless VoIP provides delay budgets for each interface in the wireless communication session from air interface 107, delay in originating network control 102, delay in transmission network 103, delay in terminating network control 104, and delay in air interface 108. There is a total cumulative delay maximum, in which, if the VoIP packet's cumulative delay exceeds that maximum, the packet is discarded instead of being delivered to the target user. Thus, great efforts are made to manage the end-to-end delay of VoIP transmissions over wireless networks.
For VoIP currently transmitted over wireless networks, robust header compression (ROHC) is typically used to compress the realtime protocol (RTP)/user datagram protocol (UDP)/IP headers for VoIP vocoded packets for capacity improvements over the air-interface. For mobile-to-mobile calls, there is a ROHC compressor at the originating mobile and there is a ROHC de-compressor at the originating RAN. The VoIP media packets from the source RAN will have uncompressed RTP/UDP/IP headers when sent from the originating to the terminating network.
From an end-to-end delay perspective, the mobile-to-mobile end-to-end VoIP media latency is the largest when compared to the mobile-to-land and land-to-mobile media latency. Here, “land” calls refer to the publicly switched telephone network (PSTN). There are several unknowns when choosing or determining media access control (MAC) parameters that affect the air-interface delay. One unknown is the loading at the source air-interface. For example, the target RAN, when receiving the VoIP packets, uses a certain delay bound or window that is calculated on a worst-case delay from the originating network. The worst-case delay is used because the target RAN does not know the cumulative delay experienced at the source RAN. Another unknown is the loading at the target air-interface. The delay margin or budget for VoIP media originating from the source network air-interface is chosen to reflect a fully loaded cell at the target RAN. Hence, the delay budget, to account for the air-interface delay at the source network, is set to a conservative value. Because of the conservative estimates used to account for these unknown parameters, a certain amount of inefficiency is built into mobile VoIP provisioning.
In VoIP services, because the VoIP users are sensitive to delay, the packet scheduler will typically throw away packets that are deemed too old. When implementing VoIP services over wireless networks as described above, because the worst-case delays are assumed when assigning delay budgets for the VoIP packets over the wireless interface, some of these packets may be discarded as being too old when, in fact, they are not. Thus, current networks that use these worst-case delay parameters may suffer in quality when perfectly valid packets are being discarded because the conservative estimate for delay pushes their age over the maximum allowable delay for that particular delay segment or hop.